Reason for revert: Reverting all CLs related to moving the eventlog, as they break Chromium tests. Original issue's description: > Move RtcEventLog object from inside VoiceEngine to Call. > > In addition to moving the logging object itself, this also moves the interface from PeerConnectionFactory to PeerConnection, which makes more sense for this functionality. An API parameter to set an upper limit to the size of the logfile is introduced. > The old interface on PeerConnectionFactory is not removed in this CL, because it is called from Chrome, it will be removed after Chrome is updated to use the PeerConnection interface. > > BUG=webrtc:4741,webrtc:5603,chromium:609749 > R=solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org, tommi@webrtc.org > > Committed: https://crrev.com/1895526c6130e3d0e9b154f95079b8eda7567016 > Cr-Commit-Position: refs/heads/master@{#13321} TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:4741,webrtc:5603,chromium:609749 Review-Url: https://codereview.webrtc.org/2111813002 Cr-Commit-Position: refs/heads/master@{#13340}
136 lines
5.3 KiB
C++
136 lines
5.3 KiB
C++
/*
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* Copyright 2011 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_API_PEERCONNECTIONFACTORY_H_
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#define WEBRTC_API_PEERCONNECTIONFACTORY_H_
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#include <memory>
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#include <string>
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#include "webrtc/api/mediacontroller.h"
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#include "webrtc/api/mediastreaminterface.h"
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#include "webrtc/api/peerconnectioninterface.h"
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#include "webrtc/base/scoped_ref_ptr.h"
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#include "webrtc/base/thread.h"
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#include "webrtc/base/rtccertificategenerator.h"
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#include "webrtc/pc/channelmanager.h"
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namespace rtc {
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class BasicNetworkManager;
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class BasicPacketSocketFactory;
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}
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namespace webrtc {
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class PeerConnectionFactory : public PeerConnectionFactoryInterface {
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public:
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void SetOptions(const Options& options) override {
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options_ = options;
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}
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// Deprecated, use version without constraints.
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rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
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const PeerConnectionInterface::RTCConfiguration& configuration,
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const MediaConstraintsInterface* constraints,
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std::unique_ptr<cricket::PortAllocator> allocator,
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
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PeerConnectionObserver* observer) override;
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virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
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const PeerConnectionInterface::RTCConfiguration& configuration,
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std::unique_ptr<cricket::PortAllocator> allocator,
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
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PeerConnectionObserver* observer) override;
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bool Initialize();
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rtc::scoped_refptr<MediaStreamInterface>
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CreateLocalMediaStream(const std::string& label) override;
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virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
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const cricket::AudioOptions& options) override;
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// Deprecated, use version without constraints.
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rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
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const MediaConstraintsInterface* constraints) override;
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virtual rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
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cricket::VideoCapturer* capturer) override;
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// This version supports filtering on width, height and frame rate.
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// For the "constraints=null" case, use the version without constraints.
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// TODO(hta): Design a version without MediaConstraintsInterface.
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// https://bugs.chromium.org/p/webrtc/issues/detail?id=5617
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rtc::scoped_refptr<VideoTrackSourceInterface> CreateVideoSource(
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cricket::VideoCapturer* capturer,
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const MediaConstraintsInterface* constraints) override;
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rtc::scoped_refptr<VideoTrackInterface> CreateVideoTrack(
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const std::string& id,
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VideoTrackSourceInterface* video_source) override;
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rtc::scoped_refptr<AudioTrackInterface>
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CreateAudioTrack(const std::string& id,
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AudioSourceInterface* audio_source) override;
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bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) override;
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void StopAecDump() override;
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bool StartRtcEventLog(rtc::PlatformFile file) override {
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return StartRtcEventLog(file, -1);
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}
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bool StartRtcEventLog(rtc::PlatformFile file,
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int64_t max_size_bytes) override;
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void StopRtcEventLog() override;
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virtual webrtc::MediaControllerInterface* CreateMediaController(
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const cricket::MediaConfig& config) const;
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virtual rtc::Thread* signaling_thread();
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virtual rtc::Thread* worker_thread();
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virtual rtc::Thread* network_thread();
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const Options& options() const { return options_; }
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protected:
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PeerConnectionFactory();
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PeerConnectionFactory(
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rtc::Thread* network_thread,
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rtc::Thread* worker_thread,
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rtc::Thread* signaling_thread,
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AudioDeviceModule* default_adm,
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const rtc::scoped_refptr<webrtc::AudioDecoderFactory>&
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audio_decoder_factory,
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cricket::WebRtcVideoEncoderFactory* video_encoder_factory,
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cricket::WebRtcVideoDecoderFactory* video_decoder_factory);
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virtual ~PeerConnectionFactory();
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private:
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cricket::MediaEngineInterface* CreateMediaEngine_w();
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bool owns_ptrs_;
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bool wraps_current_thread_;
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rtc::Thread* network_thread_;
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rtc::Thread* worker_thread_;
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rtc::Thread* signaling_thread_;
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Options options_;
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// External Audio device used for audio playback.
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rtc::scoped_refptr<AudioDeviceModule> default_adm_;
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rtc::scoped_refptr<AudioDecoderFactory> audio_decoder_factory_;
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std::unique_ptr<cricket::ChannelManager> channel_manager_;
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// External Video encoder factory. This can be NULL if the client has not
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// injected any. In that case, video engine will use the internal SW encoder.
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std::unique_ptr<cricket::WebRtcVideoEncoderFactory> video_encoder_factory_;
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// External Video decoder factory. This can be NULL if the client has not
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// injected any. In that case, video engine will use the internal SW decoder.
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std::unique_ptr<cricket::WebRtcVideoDecoderFactory> video_decoder_factory_;
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std::unique_ptr<rtc::BasicNetworkManager> default_network_manager_;
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std::unique_ptr<rtc::BasicPacketSocketFactory> default_socket_factory_;
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};
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} // namespace webrtc
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#endif // WEBRTC_API_PEERCONNECTIONFACTORY_H_
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